Hi,I noticed the calling party is behind NAT.Maybe both yate and calling party are behind different NATs?

Check the audio on sip only:Create the following rules in regexroute:^12345$=tone/dial^123456$=conf/;echo=trueCall the numbers from sip.For the first number you should hear a dial tone. This will check audio from yate to calling party.The second rule will create a conference with echo: it will echo the sound from calling party back. This way you can check audio in both ways.